JP2004508598A - Audio signal compression - Google Patents
Audio signal compression Download PDFInfo
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- JP2004508598A JP2004508598A JP2002525655A JP2002525655A JP2004508598A JP 2004508598 A JP2004508598 A JP 2004508598A JP 2002525655 A JP2002525655 A JP 2002525655A JP 2002525655 A JP2002525655 A JP 2002525655A JP 2004508598 A JP2004508598 A JP 2004508598A
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- 238000007906 compression Methods 0.000 title claims abstract description 58
- 230000006835 compression Effects 0.000 title claims abstract description 58
- 230000005236 sound signal Effects 0.000 title claims abstract description 45
- 238000000034 method Methods 0.000 claims abstract description 15
- 230000003044 adaptive effect Effects 0.000 claims description 6
- 230000005540 biological transmission Effects 0.000 claims description 3
- 238000007493 shaping process Methods 0.000 claims 1
- 238000012545 processing Methods 0.000 description 7
- 108010076504 Protein Sorting Signals Proteins 0.000 description 5
- 238000013144 data compression Methods 0.000 description 5
- 238000013459 approach Methods 0.000 description 4
- 238000010586 diagram Methods 0.000 description 4
- 238000013461 design Methods 0.000 description 3
- 230000006837 decompression Effects 0.000 description 2
- 230000001419 dependent effect Effects 0.000 description 2
- 230000006978 adaptation Effects 0.000 description 1
- 238000011156 evaluation Methods 0.000 description 1
- 238000001228 spectrum Methods 0.000 description 1
- 238000012546 transfer Methods 0.000 description 1
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- H—ELECTRICITY
- H03—ELECTRONIC CIRCUITRY
- H03M—CODING; DECODING; CODE CONVERSION IN GENERAL
- H03M7/00—Conversion of a code where information is represented by a given sequence or number of digits to a code where the same, similar or subset of information is represented by a different sequence or number of digits
- H03M7/30—Compression; Expansion; Suppression of unnecessary data, e.g. redundancy reduction
- H03M7/3002—Conversion to or from differential modulation
-
- H—ELECTRICITY
- H03—ELECTRONIC CIRCUITRY
- H03M—CODING; DECODING; CODE CONVERSION IN GENERAL
- H03M3/00—Conversion of analogue values to or from differential modulation
- H03M3/30—Delta-sigma modulation
- H03M3/458—Analogue/digital converters using delta-sigma modulation as an intermediate step
- H03M3/478—Means for controlling the correspondence between the range of the input signal and the range of signals the converter can handle; Means for out-of-range indication
- H03M3/48—Means for controlling the correspondence between the range of the input signal and the range of signals the converter can handle; Means for out-of-range indication characterised by the type of range control, e.g. limiting
- H03M3/482—Means for controlling the correspondence between the range of the input signal and the range of signals the converter can handle; Means for out-of-range indication characterised by the type of range control, e.g. limiting by adapting the quantisation step size
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
-
- G—PHYSICS
- G11—INFORMATION STORAGE
- G11B—INFORMATION STORAGE BASED ON RELATIVE MOVEMENT BETWEEN RECORD CARRIER AND TRANSDUCER
- G11B20/00—Signal processing not specific to the method of recording or reproducing; Circuits therefor
- G11B20/00007—Time or data compression or expansion
-
- G—PHYSICS
- G11—INFORMATION STORAGE
- G11B—INFORMATION STORAGE BASED ON RELATIVE MOVEMENT BETWEEN RECORD CARRIER AND TRANSDUCER
- G11B20/00—Signal processing not specific to the method of recording or reproducing; Circuits therefor
- G11B20/10—Digital recording or reproducing
- G11B20/10527—Audio or video recording; Data buffering arrangements
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04B—TRANSMISSION
- H04B14/00—Transmission systems not characterised by the medium used for transmission
- H04B14/02—Transmission systems not characterised by the medium used for transmission characterised by the use of pulse modulation
- H04B14/06—Transmission systems not characterised by the medium used for transmission characterised by the use of pulse modulation using differential modulation, e.g. delta modulation
- H04B14/062—Transmission systems not characterised by the medium used for transmission characterised by the use of pulse modulation using differential modulation, e.g. delta modulation using delta modulation or one-bit differential modulation [1DPCM]
- H04B14/064—Transmission systems not characterised by the medium used for transmission characterised by the use of pulse modulation using differential modulation, e.g. delta modulation using delta modulation or one-bit differential modulation [1DPCM] with adaptive feedback
-
- G—PHYSICS
- G11—INFORMATION STORAGE
- G11B—INFORMATION STORAGE BASED ON RELATIVE MOVEMENT BETWEEN RECORD CARRIER AND TRANSDUCER
- G11B20/00—Signal processing not specific to the method of recording or reproducing; Circuits therefor
- G11B20/00007—Time or data compression or expansion
- G11B2020/00014—Time or data compression or expansion the compressed signal being an audio signal
- G11B2020/00065—Sigma-delta audio encoding
-
- H—ELECTRICITY
- H03—ELECTRONIC CIRCUITRY
- H03M—CODING; DECODING; CODE CONVERSION IN GENERAL
- H03M3/00—Conversion of analogue values to or from differential modulation
- H03M3/30—Delta-sigma modulation
- H03M3/39—Structural details of delta-sigma modulators, e.g. incremental delta-sigma modulators
- H03M3/402—Arrangements specific to bandpass modulators
- H03M3/404—Arrangements specific to bandpass modulators characterised by the type of bandpass filters used
- H03M3/406—Arrangements specific to bandpass modulators characterised by the type of bandpass filters used by the use of a pair of integrators forming a closed loop
-
- H—ELECTRICITY
- H03—ELECTRONIC CIRCUITRY
- H03M—CODING; DECODING; CODE CONVERSION IN GENERAL
- H03M3/00—Conversion of analogue values to or from differential modulation
- H03M3/30—Delta-sigma modulation
- H03M3/39—Structural details of delta-sigma modulators, e.g. incremental delta-sigma modulators
- H03M3/412—Structural details of delta-sigma modulators, e.g. incremental delta-sigma modulators characterised by the number of quantisers and their type and resolution
- H03M3/422—Structural details of delta-sigma modulators, e.g. incremental delta-sigma modulators characterised by the number of quantisers and their type and resolution having one quantiser only
- H03M3/43—Structural details of delta-sigma modulators, e.g. incremental delta-sigma modulators characterised by the number of quantisers and their type and resolution having one quantiser only the quantiser being a single bit one
-
- H—ELECTRICITY
- H03—ELECTRONIC CIRCUITRY
- H03M—CODING; DECODING; CODE CONVERSION IN GENERAL
- H03M3/00—Conversion of analogue values to or from differential modulation
- H03M3/30—Delta-sigma modulation
- H03M3/39—Structural details of delta-sigma modulators, e.g. incremental delta-sigma modulators
- H03M3/436—Structural details of delta-sigma modulators, e.g. incremental delta-sigma modulators characterised by the order of the loop filter, e.g. error feedback type
- H03M3/438—Structural details of delta-sigma modulators, e.g. incremental delta-sigma modulators characterised by the order of the loop filter, e.g. error feedback type the modulator having a higher order loop filter in the feedforward path
- H03M3/452—Structural details of delta-sigma modulators, e.g. incremental delta-sigma modulators characterised by the order of the loop filter, e.g. error feedback type the modulator having a higher order loop filter in the feedforward path with weighted feedforward summation, i.e. with feedforward paths from more than one filter stage to the quantiser input
-
- H—ELECTRICITY
- H03—ELECTRONIC CIRCUITRY
- H03M—CODING; DECODING; CODE CONVERSION IN GENERAL
- H03M3/00—Conversion of analogue values to or from differential modulation
- H03M3/30—Delta-sigma modulation
- H03M3/458—Analogue/digital converters using delta-sigma modulation as an intermediate step
- H03M3/478—Means for controlling the correspondence between the range of the input signal and the range of signals the converter can handle; Means for out-of-range indication
- H03M3/488—Means for controlling the correspondence between the range of the input signal and the range of signals the converter can handle; Means for out-of-range indication using automatic control
- H03M3/49—Means for controlling the correspondence between the range of the input signal and the range of signals the converter can handle; Means for out-of-range indication using automatic control in feedback mode, i.e. by determining the range to be selected from one or more previous digital output values
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- Multimedia (AREA)
- Theoretical Computer Science (AREA)
- Computer Networks & Wireless Communication (AREA)
- Health & Medical Sciences (AREA)
- Computational Linguistics (AREA)
- Audiology, Speech & Language Pathology (AREA)
- Human Computer Interaction (AREA)
- Physics & Mathematics (AREA)
- Acoustics & Sound (AREA)
- Compression, Expansion, Code Conversion, And Decoders (AREA)
- Signal Processing For Digital Recording And Reproducing (AREA)
Abstract
所与のオーディオ信号を圧縮する際に得られる圧縮利得を推定する方法は、所与のオーディオ信号の選択される周波数帯域における信号パワーを抽出し、抽出された信号パワーとの相関により圧縮利得の推定を得る。A method of estimating the compression gain obtained when compressing a given audio signal is to extract the signal power in a selected frequency band of the given audio signal and to calculate the compression gain by correlation with the extracted signal power. Get an estimate.
Description
【0001】
本発明は、オーディオ信号圧縮に関する。
【0002】
国際特許出願WO98/16014は、オーディオ信号をデータ圧縮するデータ圧縮装置を開示する。このデータ圧縮装置は、オーディオ信号を受信する入力端末と、オーディオ信号をA/D変換してビットストリームを得る1−ビットA/D変換器と、ビットストリーム信号にロスレスデータ圧縮ステップを行い、データ圧縮されたビットストリーム信号を得るロスレスコーダと、データ圧縮されたビットストリーム信号を供給する出力端末とを含む。更に、データ圧縮装置を含む記録装置及び送信器装置が開示される。更に、データ圧縮装置により供給されるデータ圧縮されたビットストリーム信号をデータ伸張するデータ伸張装置と、データ伸張装置を有する再生装置及び受信器装置も開示される。
【0003】
本発明は、オーディオ圧縮の際の有利な圧縮利得推定を提供することを目的とする。このために、本発明は、独立項に記載するような圧縮利得を推定する方法及び装置、記録装置、及び、送信器を提供する。有利な実施例は、従属項に記載する。
【0004】
圧縮利得推定は、例えば、一篇の音楽といったオーディオ信号列のオーサリング及び/又は編集に使用することができる。圧縮利得推定は、圧縮符号化による圧縮比の適応制御といった信号処理パラメータの制御に使用して、記憶又は記録媒体上に格納又は記録可能なデータ量の迅速な推定を与えることにより、スーパオーディオコンパクトディスク(SACD)といった記録媒体の有限の最大記憶コンテンツ内に、例えば、一篇の音楽のコンテンツを記録することを可能にする。
【0005】
理論的には、このような推定を得るには、一篇の音楽といったオーディオ信号の一部に対する平均の実際の圧縮比又は符号化利得を、全体のオーディオ信号列に対する圧縮比の指示として使用することが考えられる。しかし、このような取り組み方法では、一篇の音楽の形である典型的なオーディオ信号列は、かなり短い時間の相関で、かなり広い圧縮比又は符号化利得の変動を有する。そのため、所望されるレベルの精度又は正確さを有する圧縮比推定を得るには、信号列の非常に多くの部分を使用しなければならない。計算時間の要件により、このような解決策は、実際には容認されないであろう。
【0006】
本発明の第1の面によると、得られる圧縮利得の有利な推定は、オーディオ信号の選択される周波数帯域に含まれる信号パワーを抽出し、その推定を、抽出される信号パワーとの相関により得る。本発明の第1の面は、圧縮比又は符号化利得の迅速な推定は、オーディオ信号の信号パワーと圧縮利得との間の相関を使用することにより入手可能であるという見識に基づいている。この面は、例えば、ダイレクトストリームデジタル形式の1ビットのビットストリーム信号をロスレスに圧縮する際に特に有利である。
【0007】
尚、標準的なPCM符号化から、得られる符号化利得は、信号パワーにおける構造(平らさ)の程度に直接関連することが周知である。しかし、本発明の上述の面では、信号パワー自体を使用し、構造は使用しない。
【0008】
以下において、本発明を、添付図面を参照しながら詳細に説明する。
【0009】
ロスレスエンコーダを調べてみると、例えば、20kHzまでのオーディオ信号帯域自体では、圧縮比に対し比較的平らな応答曲線が、信号パワーの関数として存在し、信号パワーspに依存する圧縮利得cgのかなりスティープな応答は、オーディオ信号帯域外の信号帯域に対し得られることが示される。このことは、例えば、20乃至50kHzの信号帯域に対しての図1に示す。
【0010】
図2のブロック図に示す信号処理装置では、この際立った相関を使用して、ロスレス符号化により得られる圧縮比又は符号化利得の推定を得る。アナログでもデジタルであってもよい入力オーディオ信号は、信号プロセッサ1に供給され、ここでは、入力オーディオ信号は、パラメータ制御装置2により決定される少なくとも1つの可変処理パラメータに依存して処理される。図示する例では、制御装置2により生成される制御信号は、適応シグマデルタ変調器3の適応を制御し、変調器3は、1ビットのビットストリーム信号といったいわゆるDSD(ダイレクトストリームデジタル)形式のデータストリーム信号を形成し、これは、信号プロセッサ1からの出力信号としてロスレスエンコーダ4に供給される。エンコーダ4内で得られる圧縮利得は、信号プロセッサ1の制御に使用される可変パラメータの変動によって著しく影響を受け、これは、結果として、図2に示すDVDディスク5へのSACD記録といったような記録媒体によって記録又は格納され得るデータ量に強く影響を与え得る。「Audio signal processing」なる名称で本願と同一の優先日を有する本出願人の同時係属出願を参照されたい。
【0011】
本発明の実施例によると、DSD形式のビットストリーム信号の信号パワーと、圧縮利得との間の図1に示すような相関は、符号化利得の迅速且つ正確な推定を与えるよう使用される。図示するように、信号パワーは、信号プロセッサ1からの出力に接続される抽出及び相関装置6により、ビットストリーム信号から抽出され、圧縮比又は符号化利得推定は、入力制御信号としてパラメータ制御装置2に供給される。
【0012】
図示するように、パラメータ制御される信号プロセッサ1の構成素子は、一般的に、適応シグマデルタ変調器3である。幾つかの取り組み方法が、シグマデルタ変調器の構造の適応又は変更を行うために使用可能である。例えば、異なるオーダを有するシグマデルタ変調器の構造間でシフトする、又は、変調器の高周波雑音において構造を形成するといった取り組み方法がある。
【0013】
従って、第3のオーダの、第5のオーダの、又は、第7のオーダの変調器構造間でシフトすることにより、圧縮比は、一般的に、第3のオーダの構造に対する約3.7から、第7のオーダの構造に対するたった2.3又はそれ以下に亘る範囲で得られ、このことは、エンコーダ4からの圧縮されたビットストリーム信号のデータ量を、例えば、DVDディスク5といった記録媒体の限られた記録又は記憶容量にフィットするよう整形するために使用される。
【0014】
図3は、図2に示す信号プロセッサ1に使用する第5のオーダのシグマデルタ変調器の好適なトポロジーを示す。図示するトポロジーは、多数の共振器構造に基づいており、共振器R1、R2、…、R5のフィードバックループにおける係数c1、c2、…、c5が、ループフィルタの極(又は、雑音伝達関数のゼロ)を決定する。図示するトポロジーは第5のオーダの変調器に対するものであるが、同一のトポロジーを、単にもう1つの共振器構造を付加することにより第7のオーダの変調器に使用してもよい。
【0015】
シグマデルタ変調器の通常の設計では、極は通常オーディオ帯域内に配置される。しかし、本発明では、少なくとも1つの極を、オーディオ帯域外に配置し、シグマデルタスペクトルの高周波部(それ以外は略平らである)に追加の構造を形成することが好適である。
【0016】
従って、シグマデルタ変調器の標準設計では、極は、一般的に8.7、15.7、及び、19.5IkHzに配置されるが、本発明では、最後の極は、20kHz領域から高い周波数領域、例えば、300kHz以上の周波数にシフトすることが好適であり、これは、圧縮利得のかなりの増加につながる。このことは、信号対雑音のパフォーマンスを僅かに低下させる場合があるが、これは、通常は、全く容認可能であり、というのは、余分の雑音は、人間の耳にはあまり聞こえない高い周波帯域に発生するからである。
【0017】
極位置を、20kHz領域から高周波にシフトすることは、現行の変調器構造に、例えば、低域通過ループフィルタに並列して、別個の追加の帯域通過フィルタを付加することにより達成される。2次バターワース(Butterworth)帯域通過フィルタを上述したような並列フィルタとして使用することにより、かなりの圧縮利得の増加が実現され、変調器は大きい入力に対し依然として安定し、オーディオ帯域における信号対雑音のパフォーマンスは依然として変更されていない変調器と比べても実質的に変化しない。
【0018】
異なるシグマデルタ変調器構造間でシフトすることは、ビットストリーム信号の一部の周波数帯域における信号パワーと、圧縮比又は符号化利得との間の関係に変動をもたらす場合があり、従って、圧縮比推定のための基礎として使用する相関に影響を与える場合があるという事実を考慮することにより、相関を固定するために最初にキャリブレーション処理を行うことが有利である。
【0019】
この目的のため、固定シグマデルタ変調器が、ロスレスエンコーダ4前の最終処理装置として組込まれる。これは、図2中、破線のブロック7に示す。従って、信号パワーは、固定シグマデルタ変調器7からロスレスエンコーダ4に供給される信号から抽出されることが好適である。
【0020】
図2に示す上述の信号プロセッサの実施例では、適応シグマデルタ変調器を使用して、ビットストリーム信号の信号パワーから得られる相関に応答して圧縮比又は符号化利得を適応する又は変更するが、固定或いは適応型のシグマデルタ変調器前で、信号プロセッサにおける帯域幅制限低域通過フィルタの組み込みの信号レベルを減少するといった他のアプローチも、別個に又は組合わせて使用してもよい。
【0021】
更に、本発明の実施例による方法を実行する信号処理装置は、後続の圧縮符号化において得られる圧縮比の推定を、例えば、オーディオ信号列の、純粋な評価、フォーマッティング、オーサリング、及び/又は編集のためのスタジオ環境に別個で使用可能である限りは、それ自体が圧縮符号化手段を含む必要はない。
【0022】
上述した実施例は、本発明を例示するものであって、制限するものではないことを理解すべきである。当業者は、特許請求の範囲から逸脱することなく多くの代替となる実施例を設計することが可能である。請求項において、括弧内に置かれる任意の参照符号は、その請求項を制限するものではない。「含む」という用語は、請求項に列挙される構成素子及び段階以外の構成素子及び段階の存在を排除するものではない。本発明は、幾つかの別個の構成素子を含むハードウェア、及び、好適にプログラムされたコンピュータによって実施可能である。装置請求項において、幾つかの手段が列挙されるが、手段のうちのいくつかは、全く同一のハードウェアアイテムとして具現化されることが可能である。特定の手段が相互に異なる従属項において記載されるという事実は、これらの手段を有利に組合わせて使用することができないということを示すものではない。
【図面の簡単な説明】
【図1】
圧縮利得とオーディオ信号の選択周波数帯域における信号パワーとの関係を示す図である。
【図2】
本発明の信号処理装置の実施例が組込まれるデジタルオーディオ信号記録又は送信チェーンを示す単純化されたブロック図である。
【図3】
図2に示す信号プロセッサに使用する第5のオーダのシグマデルタ変調器を示す単純化されたトポロジー図である。[0001]
The present invention relates to audio signal compression.
[0002]
International Patent Application WO 98/16014 discloses a data compression device for compressing an audio signal. The data compression apparatus includes an input terminal for receiving an audio signal, a 1-bit A / D converter for A / D converting the audio signal to obtain a bit stream, and performing a lossless data compression step on the bit stream signal. It includes a lossless coder that obtains a compressed bit stream signal, and an output terminal that supplies a data compressed bit stream signal. Further, a recording device and a transmitter device including a data compression device are disclosed. Furthermore, a data decompression device for decompressing a data-compressed bit stream signal supplied by the data compression device, and a reproducing device and a receiver device having the data decompression device are also disclosed.
[0003]
It is an object of the present invention to provide an advantageous compression gain estimation during audio compression. To this end, the present invention provides a method and apparatus for estimating compression gain, a recording device, and a transmitter as described in the independent claims. Advantageous embodiments are described in the dependent claims.
[0004]
The compression gain estimation can be used, for example, for authoring and / or editing an audio signal sequence, such as a piece of music. Compression gain estimation is used to control signal processing parameters, such as adaptive control of the compression ratio by compression encoding, and to provide a quick estimate of the amount of data that can be stored or recorded on a storage or recording medium, thereby providing a super audio compact For example, one piece of music content can be recorded in a finite maximum storage content of a recording medium such as a disc (SACD).
[0005]
Theoretically, to obtain such an estimate, use the average actual compression ratio or coding gain for a portion of the audio signal, such as a piece of music, as an indication of the compression ratio for the entire audio signal sequence. It is possible. However, in such an approach, a typical audio signal sequence in the form of a piece of music has a fairly wide compression ratio or coding gain variation with a fairly short time correlation. Thus, to obtain a compression ratio estimate with the desired level of precision or accuracy, a very large portion of the signal sequence must be used. Due to the computation time requirements, such a solution would not be really acceptable.
[0006]
According to a first aspect of the invention, an advantageous estimate of the resulting compression gain is to extract the signal power contained in the selected frequency band of the audio signal and to make that estimate by correlation with the extracted signal power. obtain. A first aspect of the invention is based on the insight that a quick estimation of the compression ratio or coding gain is obtainable by using a correlation between the signal power of the audio signal and the compression gain. This aspect is particularly advantageous, for example, when losslessly compressing a 1-bit bit stream signal in a direct stream digital format.
[0007]
It is well known that the coding gain obtained from standard PCM coding is directly related to the degree of structure (flatness) in signal power. However, in the above aspects of the invention, the signal power itself is used, and no structure is used.
[0008]
Hereinafter, the present invention will be described in detail with reference to the accompanying drawings.
[0009]
Looking at lossless encoders, for example, in the audio signal band itself up to 20 kHz, a relatively flat response curve to compression ratio exists as a function of signal power, and the compression gain cg, which depends on signal power sp, is quite large. It is shown that a steep response is obtained for signal bands outside the audio signal band. This is shown, for example, in FIG. 1 for a signal band of 20 to 50 kHz.
[0010]
The signal processing device shown in the block diagram of FIG. 2 uses this outstanding correlation to obtain an estimate of the compression ratio or coding gain obtained by lossless coding. An input audio signal, which may be analog or digital, is provided to a signal processor 1 where the input audio signal is processed depending on at least one variable processing parameter determined by a parameter controller 2. In the illustrated example, the control signal generated by the control device 2 controls the adaptation of the adaptive sigma-delta modulator 3, and the modulator 3 outputs so-called DSD (direct stream digital) data such as a 1-bit bit stream signal. Form a stream signal, which is supplied to the lossless encoder 4 as an output signal from the signal processor 1. The compression gain obtained in the encoder 4 is significantly affected by the variation of the variable parameters used to control the signal processor 1, which results in a recording such as a SACD recording on a DVD disk 5 shown in FIG. It can strongly affect the amount of data that can be recorded or stored by the medium. See Applicant's co-pending application having the same priority date as the present application under the name "Audio signal processing".
[0011]
According to an embodiment of the present invention, the correlation between the signal power of the bit stream signal in DSD format and the compression gain as shown in FIG. 1 is used to provide a quick and accurate estimation of the coding gain. As shown, the signal power is extracted from the bit stream signal by an extractor and correlator 6 connected to the output from the signal processor 1 and the compression ratio or coding gain estimation is performed as a parameter control 2 Supplied to
[0012]
As shown, the component of the parameter controlled signal processor 1 is typically an adaptive sigma delta modulator 3. Several approaches can be used to adapt or change the structure of the sigma-delta modulator. For example, there are approaches to shifting between structures of sigma-delta modulators having different orders, or forming structures in the high frequency noise of the modulator.
[0013]
Thus, by shifting between third-order, fifth-order, or seventh-order modulator structures, the compression ratio will generally be about 3.7 relative to the third-order structure. From the range of only 2.3 or less for the structure of the seventh order, which means that the amount of data of the compressed bit stream signal from the encoder 4 can be reduced by a recording medium such as a DVD disk 5. Used to fit the limited recording or storage capacity of
[0014]
FIG. 3 shows a preferred topology of a fifth order sigma-delta modulator for use in the signal processor 1 shown in FIG. The illustrated topology is based on a number of resonator structures, where the coefficients c1, c2, ..., c5 in the feedback loop of the resonators R1, R2, ..., R5 are the poles of the loop filter (or the zero of the noise transfer function). ). Although the topology shown is for a fifth order modulator, the same topology may be used for a seventh order modulator simply by adding another resonator structure.
[0015]
In a typical design of a sigma-delta modulator, the poles are usually located in the audio band. However, it is preferred in the present invention that the at least one pole be located outside the audio band, forming an additional structure in the high frequency portion of the sigma delta spectrum (otherwise it is generally flat).
[0016]
Thus, in the standard design of a sigma-delta modulator, the poles are typically located at 8.7, 15.7, and 19.5 IkHz, but in the present invention, the last pole is located at a higher frequency from the 20 kHz region. It is preferred to shift to a frequency in the region, for example 300 kHz or higher, which leads to a considerable increase in the compression gain. This can slightly degrade the signal-to-noise performance, but it is usually quite acceptable because extra noise is present at higher frequencies that are less audible to the human ear. This is because it occurs in the band.
[0017]
Shifting the pole position from the 20 kHz region to higher frequencies is achieved by adding a separate additional bandpass filter to the current modulator structure, for example, in parallel with a low pass loop filter. By using a second-order Butterworth bandpass filter as a parallel filter as described above, a significant increase in compression gain is achieved, the modulator is still stable for large inputs, and the signal to noise in the audio band is reduced. The performance remains substantially unchanged compared to the modulator which has not been modified.
[0018]
Shifting between different sigma-delta modulator structures may cause a variation in the relationship between signal power in some frequency bands of the bitstream signal and the compression ratio or coding gain, and thus the compression ratio It is advantageous to first perform a calibration process to fix the correlation by taking into account the fact that it may affect the correlation used as a basis for the estimation.
[0019]
For this purpose, a fixed sigma delta modulator is incorporated as a final processing device before the lossless encoder 4. This is indicated by the dashed block 7 in FIG. Therefore, the signal power is preferably extracted from the signal supplied from the fixed sigma delta modulator 7 to the lossless encoder 4.
[0020]
In the above-described signal processor embodiment shown in FIG. 2, an adaptive sigma delta modulator is used to adapt or change the compression ratio or coding gain in response to a correlation derived from the signal power of the bitstream signal. Other approaches, such as reducing the signal level of the built-in bandwidth-limited low-pass filter in the signal processor, before the fixed or adaptive sigma-delta modulator, may also be used separately or in combination.
[0021]
Furthermore, the signal processing device performing the method according to an embodiment of the present invention may provide an estimate of the compression ratio obtained in the subsequent compression coding, for example, for a pure evaluation, formatting, authoring and / or editing of the audio signal sequence. It need not itself include compression encoding means as long as it can be used separately in the studio environment for.
[0022]
It should be understood that the above-described embodiments are illustrative of the present invention and not limiting. One skilled in the art can design many alternative embodiments without departing from the scope of the claims. In the claims, any reference signs placed between parentheses shall not limit the claim. The word "comprising" does not exclude the presence of elements or steps other than those listed in a claim. The invention can be implemented by means of hardware comprising several distinct components, and by means of a suitably programmed computer. In the device claim, some means are enumerated, but some of the means may be embodied as identical hardware items. The mere fact that certain measures are recited in mutually different dependent claims does not indicate that a combination of these measures cannot be used to advantage.
[Brief description of the drawings]
FIG.
FIG. 4 is a diagram illustrating a relationship between a compression gain and a signal power in a selected frequency band of an audio signal.
FIG. 2
FIG. 2 is a simplified block diagram showing a digital audio signal recording or transmission chain into which an embodiment of the signal processing device of the present invention is incorporated.
FIG. 3
FIG. 3 is a simplified topology diagram illustrating a fifth order sigma delta modulator used in the signal processor shown in FIG. 2.
Claims (11)
上記所与のオーディオ信号の選択された周波数帯域における信号パワーを抽出する段階と、
上記抽出された信号パワーとの相関により上記圧縮利得の推定を得る段階とを含む方法。A method for estimating the compression gain obtained when compressing a given audio signal, comprising:
Extracting signal power in a selected frequency band of the given audio signal;
Obtaining an estimate of the compression gain by correlation with the extracted signal power.
上記オーディオ信号を圧縮し、圧縮されたオーディオ信号を得る段階と、
上記圧縮されたオーディオ信号を上記記録媒体に記録する段階とを含み、
上記方法は更に、
上記オーディオ信号の選択された周波数帯域における信号パワーを抽出し、上記抽出された信号パワーとの相関により圧縮利得の推定を得ることにより、上記圧縮段階において得られる圧縮利得を推定する段階と、
上記推定された圧縮利得に依存して上記圧縮段階を制御する段階とを含む方法。A method for recording an audio signal on a recording medium, comprising:
Compressing the audio signal to obtain a compressed audio signal;
Recording the compressed audio signal on the recording medium.
The method further comprises:
Estimating the compression gain obtained in the compression step by extracting signal power in a selected frequency band of the audio signal and obtaining an estimate of compression gain by correlation with the extracted signal power;
Controlling the compression step depending on the estimated compression gain.
上記オーディオ信号を圧縮し、圧縮されたオーディオ信号を得る段階と、
上記圧縮されたオーディオ信号を、伝送媒体を介し送信する段階とを含み、
上記方法は更に、
上記オーディオ信号の選択された周波数帯域における信号パワーを抽出し、上記抽出された信号パワーとの相関により圧縮利得の推定を得ることにより、上記圧縮段階において得られる圧縮利得を推定する段階と、
上記推定された圧縮利得に依存して上記圧縮段階を制御する段階とを含む方法。A method of transmitting an audio signal,
Compressing the audio signal to obtain a compressed audio signal;
Transmitting the compressed audio signal via a transmission medium,
The method further comprises:
Estimating the compression gain obtained in the compression step by extracting signal power in a selected frequency band of the audio signal and obtaining an estimate of compression gain by correlation with the extracted signal power;
Controlling the compression step depending on the estimated compression gain.
上記所与のオーディオ信号の選択された周波数帯域における信号パワーを抽出する手段と、
上記抽出された信号パワーとの相関により上記圧縮利得の推定を得る手段とを含む装置。An apparatus for estimating a compression gain obtained when compressing a given audio signal,
Means for extracting signal power in a selected frequency band of the given audio signal;
Means for obtaining an estimate of said compression gain by correlation with said extracted signal power.
上記オーディオ信号を圧縮し、圧縮されたオーディオ信号を得る手段と、
上記圧縮されたオーディオ信号を上記記録媒体に記録する手段とを含み、
上記記録装置は更に、
上記圧縮の際に得られる圧縮利得を推定する請求項9記載の装置と、
上記推定された圧縮利得に依存して上記圧縮を制御する手段とを含む装置。A recording device that records an audio signal on a recording medium,
Means for compressing the audio signal to obtain a compressed audio signal;
Means for recording the compressed audio signal on the recording medium,
The recording device further comprises:
An apparatus according to claim 9 for estimating a compression gain obtained during the compression,
Means for controlling said compression in dependence on said estimated compression gain.
上記オーディオ信号を圧縮し、圧縮されたオーディオ信号を得る手段と、
上記圧縮されたオーディオ信号を、伝送媒体を介し送信する手段とを含み、
上記送信器は更に、
上記圧縮の際に得られる圧縮利得を推定する請求項9記載の装置と、
上記推定された圧縮利得に依存して上記圧縮を制御する手段とを含む送信器。A transmitter for transmitting an audio signal,
Means for compressing the audio signal to obtain a compressed audio signal;
Means for transmitting the compressed audio signal via a transmission medium,
The transmitter further comprises:
An apparatus according to claim 9 for estimating a compression gain obtained during the compression,
Means for controlling said compression in dependence on said estimated compression gain.
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PCT/EP2001/010336 WO2002021525A1 (en) | 2000-09-08 | 2001-09-06 | Audio signal compression |
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